And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
Breaking this dependency is required for using FakePortAllocator in chromium tests to make the windows component build work.
Bug: chromium:1408420
Change-Id: I4215b92c1d1430156107605e5b054926b30f83f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291114
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#39180}
This is to avoid using 0 as a default value.
Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp.
Bug: webrtc:13609
Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118
Reviewed-by: Olov Brändström <brandstrom@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39177}
- Lowering the energy threshold for updating the accumulated error.
- Not using the pre-echo estimate in the initial frames when the matched filters have been recently initialized.
- Slight speed up for the increases in the accumulated error.
- Not periodically resetting the accumulated error.
Bug: webrtc:14205
Change-Id: Ic337332e263b27d7a3aba0ab4b371517780f9c90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291320
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39175}
This ensure transport feedback is sent for RTX packets that are received before media payload packets.
Bug: webrtc:14795, webrtc:14817
Change-Id: I6a2579b87c8863e003decb2b2559ef51a852cadb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291119
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39174}
also refactor the code to have FillSendCodecStats/FillReceiveCodecStats
methods for similarity with the video engine code.
BUG=webrtc:14808
Change-Id: Ib0687f36a4b4a71c849e0b4918e50592d7772ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290891
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39172}
so it can be used in downstream Chrome tests
BUG=None
Change-Id: I4b3e1f172e8eb2ba01ab5c257f3626223781da31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291116
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39163}
Based on offline testing; needed to allow input volume adaptations
more frequently. Note that if the estimated speech level falls in
the target range, the recommended input volume won't change and
hence the new lower threshold won't necessarily increase the
number of adjustments.
Bug: webrtc:7494
Change-Id: Iabb501c188da238ea7b7137175bcfe09239c90a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291110
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39161}
Creating a data channel or negotiating it can make the SCTP transport
name go from nothing (empty string) to something. Inside the
RTCStatsCollector this is relevant because which transports we have
affect which certificates we should cache, so this is an instance of
having to call ClearStatsCache().
The bug is that we don't. This CL fixes the bug.
I tried to create unittests to cover this, but I was unable to
reproduce the race in a testing environment (if I did it would have
hit an RTC_DCHECK). Not ideal... but I hope we can land it anyway since
the fix is trivial and clearing the cache in response to API calls is
worst case harmless.
Bug: webrtc:14844
Change-Id: Ia7174cde040839e5555237db6de285297120b123
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291112
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39160}
Certificates being missing is a sign of a bug (e.g. webrtc:14844, to be
fixed separately) which is why we have a DCHECK. But this DCHECK does
not protect against accessing the invalid iterator if it is a release
build. This CL makes that safe.
Bug: chromium:1408392
Change-Id: I97a82786028e41c58ef8ef15002c3f959bbec7f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291109
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39159}
This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae.
Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104
Original change's description:
> Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
>
> This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
>
> Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
>
>
> Original change's description:
> > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
> >
> > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> > Therefore DirectTransport is provided with the extension mapping.
> >
> > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
> >
> >
> > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> > Bug: webrtc:7135, webrtc:14795
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39137}
>
> Bug: webrtc:7135, webrtc:14795, webrtc:14833
> Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39146}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39157}
This should help to debug problems with DNS resolution of turn server
name. E.g. DNS server returns fake IP address to block turn server.
Similar change to stun_port.cc were done in
https://webrtc-review.googlesource.com/c/src/+/215926
Bug: None
Change-Id: If1cc410f4cd0f89620d3678aabf05a0f1b22a393
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290992
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#39152}
This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> It is possible to have AddTransceiver calls with an empty array
> of encodings or AddTrack calls. In both cases, before negotiation,
> the sender's encodings array would be empty and it was not possible
> to update any value.
>
> Now, a default entry should be created in those cases, allowing to
> update the configuration before negotiation.
>
> Bug: webrtc:10567
> Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> Auto-Submit: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39126}
Bug: webrtc:10567
Change-Id: I558a95f7b587302b5e95f6ec26d1eb1fedf3dbed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39150}
This reverts commit fc5d627cef71f906e921476c2e6b1e01d07732fe.
Reason for revert: Breaks upstream WPT tests
Original change's description:
> Reland "Ensure RTCRtpSenders are always created with one encoding"
>
> This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
>
> Original change's description:
> > Ensure RTCRtpSenders are always created with one encoding
> >
> > It is possible to have AddTransceiver calls with an empty array
> > of encodings or AddTrack calls. In both cases, before negotiation,
> > the sender's encodings array would be empty and it was not possible
> > to update any value.
> >
> > Now, a default entry should be created in those cases, allowing to
> > update the configuration before negotiation.
> >
> > Bug: webrtc:10567
> > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> > Auto-Submit: Florent Castelli <orphis@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39126}
>
> Bug: webrtc:10567
> Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39145}
Bug: webrtc:10567
Change-Id: If9b5adb5debb7c87a15662a8d0f232405a0e8136
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291221
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39147}
This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
Original change's description:
> Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
>
> PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> Therefore DirectTransport is provided with the extension mapping.
>
> CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
>
>
> Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> Bug: webrtc:7135, webrtc:14795
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39137}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39146}
This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> It is possible to have AddTransceiver calls with an empty array
> of encodings or AddTrack calls. In both cases, before negotiation,
> the sender's encodings array would be empty and it was not possible
> to update any value.
>
> Now, a default entry should be created in those cases, allowing to
> update the configuration before negotiation.
>
> Bug: webrtc:10567
> Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> Auto-Submit: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39126}
Bug: webrtc:10567
Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39145}
This is used by CRD and export is required for component builds
to work properly.
Bug: chromium:1291247
Change-Id: I281e490b7d00cbd074b96eac905976af38400f8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291200
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39143}
PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
Therefore DirectTransport is provided with the extension mapping.
CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
Bug: webrtc:7135, webrtc:14795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39137}
Receivers no longer need to set extensions in the configuration. That field will be removed in a follow up.
Tested with:
video_loopback --rtp_dump_name="./my.rtpdump" --duration=10
video_replay --input_file=./my.rtpdump
Bug: webrtc:14795
Change-Id: I952cd487cb2f3be8be01a90f6a2312f1fef5d93e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290995
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39136}
